Asterisk (Free PBX 10 or above) Configuration Settings as a registered sip extension
Method 1
This is the basic method that 99% of asterisk users connect. There are 2 other methods that can be conected. Some people tends to mix tthem and they all call them as sip trunks, but in reality they are all different
.
Method 1 - Registered as a sip extension
Method 2 - IP Authenticated as a sip extension
Method 3 - IP Authenticated as a sip trunk (WorldDialPoint customers can only acess this information)
(Method 2 and 3 can be found in our knowledbase and coverd in detail)
Attention: If you have mulitple Phone numbers on the same extension, with this method you will NOT be able to differentiate the incoming calls. You will need to use Method 3.or add more extensions.
Here is what is required. (where xxxxxxxxx is your login number that you received in your last email)
FreePBX 10 setttings FPBX-2.10.0(1.8.13.0) but should work with earlier versions
type=peer
host=VOIP_IP_SERVER (received in the activation email)
qualify=yes
canreinvite=no
username=xxxxxxxxx
fromuser=xxxxxxxxx
secret=PASSWORD
insecure=port,invite
disallow=all
allow=alaw
-----------------------------------
Incoming
xxxxxxxxx
username=xxxxxxxxx
type=user
secret=PASSWORD
qualify=no
insecure=very
fromuser=xxxxxxxxx
context=from-trunk
canreinvite=no
xxxxxxxxx:PASSWORD@VOIP_IP_SERVER/xxxxxxxxx
With the above config, you need to have the phones also use G711a codec.
Trustrpid - NO
Sendrpid - YES
ii) Don't forget the outbound rules.
iii) Lastly set the SIP extensions of your phones with the CLI in E164 format from the numbers you have with us (if this is not done, the carriers will block the calls as INVALID NUMBER.
Please Note this info is supplied as is and we are not liable for any loss or hacking to your Asterisk. If you do not know how to secure properly an Asterisk Installation please contact a Proffessional to do it or contact us and we can reffer you to a Proffessional Installer.
